Extract Audio from CAVS to AU — Free Online Tool
Extract audio from CAVS (Chinese Audio Video Standard) files and save it as AU format using PCM 16-bit big-endian encoding. This tool strips the video stream entirely and outputs a lossless, uncompressed PCM audio file compatible with Unix systems and classic Sun/NeXT audio workflows.
to
FFmpeg Command
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Note: Browser-based encoding uses approximate quality targets. For precise CRF compression, copy the FFmpeg command above and run it on your desktop.
Estimated output:
Conversion Complete!
DownloadHow It Works
CAVS files typically carry AAC-encoded audio paired with video encoded using libx264. During this conversion, FFmpeg discards the video stream entirely and decodes the AAC audio from the CAVS container. That decoded audio is then re-encoded as PCM signed 16-bit big-endian (pcm_s16be), which is the default and most compatible codec for the AU format. The result is an uncompressed waveform stored in Sun's AU container — a format with a minimal header and raw PCM data. Because AAC is lossy, the output AU file is not a lossless representation of an original source, but it is an uncompressed representation of the audio as it existed in the CAVS file, with no further quality loss introduced by this conversion step.
What Each Flag Does
| Flag | What it does |
|---|---|
ffmpeg
|
Invokes the FFmpeg tool, which handles all media demuxing, decoding, and encoding. In this browser-based tool, it runs as FFmpeg.wasm compiled to WebAssembly, executing locally in your browser with no server involvement. |
-i input.cavs
|
Specifies the input CAVS file. FFmpeg reads and demuxes the CAVS container, identifying the libx264 video stream and AAC audio stream inside it for further processing. |
-vn
|
Disables video output entirely, instructing FFmpeg to ignore the libx264 video stream from the CAVS file. This is essential because AU is a pure audio format and cannot contain video data. |
-c:a pcm_s16be
|
Sets the audio codec to PCM signed 16-bit big-endian, which decodes the AAC audio from the CAVS file and re-encodes it as uncompressed, raw PCM samples in big-endian byte order — the native and default codec for the Sun AU format. |
output.au
|
Specifies the output filename with the .au extension, causing FFmpeg to write the extracted uncompressed audio into a Sun AU container with its characteristic minimal header and raw pcm_s16be audio data. |
Common Use Cases
- Extracting audio tracks from CAVS broadcast recordings for archival on Unix or Linux systems that natively support the AU format
- Preparing audio from Chinese digital broadcast content for analysis or processing in legacy Unix-based audio tools like SoX or AU-aware DAW plugins
- Stripping audio from CAVS video files for use in Sun Microsystems or NeXT workstation software environments that expect PCM AU files
- Converting CAVS broadcast audio to an uncompressed PCM AU file before feeding it into a pipeline that requires raw big-endian audio data
- Isolating speech or commentary tracks from CAVS-encoded Chinese television content for transcription or linguistic research workflows
- Generating AU audio files from CAVS sources to embed or stream in early web or Java applet contexts that relied on the AU format for cross-platform audio
Frequently Asked Questions
The CAVS file contains AAC audio, which is a lossy codec, so some quality was already lost when the CAVS file was originally created. This conversion decodes that AAC audio and stores it as uncompressed PCM in the AU file, introducing no additional quality loss. The output AU file is an uncompressed snapshot of the audio exactly as it was encoded in the CAVS source — not a restoration of the original pre-AAC audio.
The AU format was developed by Sun Microsystems for Unix systems, which historically used big-endian byte order — hence pcm_s16be (signed 16-bit big-endian) is the natural and most widely compatible codec for AU files. While AU technically supports other encodings like pcm_alaw, pcm_mulaw, and pcm_s8, pcm_s16be provides the best balance of fidelity and compatibility with software that reads AU files. FFmpeg selects it automatically as the default AU codec.
Significantly larger. The audio in a CAVS file is AAC-compressed, typically at 128k bitrate, which is highly space-efficient. The AU output uses uncompressed PCM at 16 bits per sample, so for stereo audio at 44.1kHz, the data rate is approximately 1.41 Mbps — roughly 10 to 11 times more data than 128k AAC. A 10-minute CAVS audio track might produce an AU file of around 100MB where the compressed audio in the CAVS was only 9–10MB.
Yes. The AU format supports several PCM variants. To use mu-law encoding (common in telephony), modify the FFmpeg command to: ffmpeg -i input.cavs -vn -c:a pcm_mulaw output.au. You can also use pcm_alaw, pcm_s8, or pcm_u8 by substituting the codec name after -c:a. Note that pcm_mulaw and pcm_alaw are lossy encodings suited for voice, while pcm_s16be is the best choice for music or general audio.
Generally no. The AU format has an extremely minimal header — it was designed by Sun Microsystems for simplicity and contains only basic technical parameters like sample rate, channel count, and encoding type. It has no standardized metadata fields for tags like title, artist, or album. Any metadata stored in the CAVS container will be lost during this conversion, which is expected behavior for AU output.
On Linux or macOS, you can use a shell loop: for f in *.cavs; do ffmpeg -i "$f" -vn -c:a pcm_s16be "${f%.cavs}.au"; done. On Windows Command Prompt, use: for %f in (*.cavs) do ffmpeg -i "%f" -vn -c:a pcm_s16be "%~nf.au". This runs the same extraction command on every CAVS file in the current directory and saves each result as a matching AU file. This is particularly useful for processing large batches of CAVS broadcast recordings.
Technical Notes
CAVS (Chinese Audio Video Standard) is a Chinese national broadcast format that uses libx264 for video and AAC for audio, making it a somewhat conventional codec pairing despite its specialized container. The AU format, by contrast, is one of the oldest and simplest audio containers in existence — developed by Sun Microsystems in the late 1980s for Unix workstations. Its header contains only an offset to the data block, data size, encoding type, sample rate, and channel count, with optional space for an ASCII annotation. This simplicity means AU files are extremely portable across Unix environments but carry no modern metadata support. The pcm_s16be codec used here produces full CD-quality audio when the source sample rate matches 44.1kHz or 48kHz, but note that AU files with non-standard sample rates may not play correctly in all applications. Since CAVS audio is AAC at a lossy 128k default, the decoded PCM in the AU file represents the post-lossy-compression waveform — bit-perfect reconstruction of the original pre-encoding audio is not possible. The -vn flag is critical here: without it, FFmpeg would attempt to encode video into the AU container, which would fail immediately since AU is an audio-only format. The conversion runs entirely client-side via FFmpeg.wasm, so no CAVS content is transmitted to any server.